Digitized speech sounds consume relatively large amounts of signal bandwidth. Accordingly, telecommunications systems employ various data compression or "speech coding" schemes to convert speech sounds into codes which consume comparatively small amounts of signal bandwidth. Instead of transmitting the original speech sounds, or their digitized equivalents, the system transmits only the codes to a remote receiver which decodes them to reproduce the original speech sounds. The system thus conserves the available transmission bandwidth, making it possible to simultaneously transmit larger volumes of speech sounds, without resorting to an expensive increase in bandwidth. The prior art has evolved a variety of speech coding techniques, all having the objective of minimizing the information which must pass from the transmitter to the receiver, while enabling the receiver to faithfully reproduce the original speech sounds.
State of the art speech coding techniques typically employ a transmitter and a receiver having identical filters and identical "excitation codebooks". The excitation codebooks contain a variety of prestored waveform shapes or "codevectors", each codevector consisting of a plurality of samples. The codevectors are used to excite the filters, to which periodically updated filtration parameters are applied, thereby enabling the filters to model changes in a speaker's vocal tract. The filters output reconstructed speech vectors which are compared with the input speech sound vectors to select the reconstructed speech vectors which most closely approximate the original speech.
At the transmitter, series of previously reconstructed speech vectors are periodically compared to the input speech vectors, to select the codevector sequence which yields the best reconstructed speech vector. The transmitter sends to the receiver a sequence of codebook indices, which represent the locations of the selected codevectors within the codebook, together with the filtration parameters which were applied to the transmitter's filter while the codevectors were selected. The receiver uses the received sequence of codebook indices to recover the selected codevectors from its own codebook, decodes the transmitted filtration parameters and applies them to its own filter, then passes the recovered codevectors through the filter to yield a sequence of reconstructed speech vectors which reproduce the original speech sounds.
The present invention improves upon the prior art speech coding technique aforesaid by eliminating the need to transmit the filtration parameters to the receiver. Only the codebook indices are transmitted. The transmitter and the receiver apply a backward predictive analysis technique to previously recovered codevectors to derive the required filtration parameters.